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244915US8 



TITLE OF THE INVENTION 

METHOD, APPARATUS, AND SYSTEM FOR FEATHERING DATA IN A 

COMMUNICATION NETWORK 

CROSS REFERENCE TO RELATED APPLICATIONS 
The present application contains subject matter related to that disclosed in 
commonly owned U.S. patent application Serial No. [to be determined], attorney 
docket #24491 8US8, entitled METHOD, APPARATUS, AND SYSTEM FOR 
DOWNSTREAM RECOVERY IN A COMMUNICATION NETWORK, filed 
November 26, 2003; U.S. patent application Serial No. [to be determined], attorney 
docket #244917US8, entitled METHOD, APPARATUS, AND SYSTEM FOR 
USING A SYNCHRONOUS BURST TIME PLAN IN A COMMUNICATION 
NETWORK, filed November 26, 2003; U.S. patent application Serial No. [to be 
determined], attorney docket #244906US8, entitled METHOD, APPARATUS, AND 
SYSTEM FOR DEMAND ASSIGNMENT IN A COMMUNICATION NETWORK, 
filed November 26, 2003; and U.S. patent application Serial No. [to be determined], 
attorney docket #242845US8, entitled METHOD, APPARATUS, AND SYSTEM 
FOR CALCULATING AND MAKING A SYNCHRONOUS BURST TIME PLAN 
IN A COMMUNICATION NETWORK, filed November 26, 2003. 

BACKGROUND OF THE INVENTION 

Field of the Invention 

This invention relates in general to satellite communication networks and in 
particular to a method, apparatus and system for efficiently optimizing bandwidth and 
sharing resources between multiple users of a satellite communication network. 



Discussion of the Background 

Geo-synchronous satellite communication networks have existed for decades 
in various topologies and using various methods for sharing a fixed bandwidth 
channel between multiple users (Pritchard, Wilbur L., and Joseph A. Scivlli, Satellite 
5 Communication Svste ms Engineering. Prentice-Hall, 1 986, incorporated in its entirely 
herein by reference). As these networks evolved. Time Division Multiple Access 
(TDMA) and Frequency Division Multiple Access (FDMA) bandwidth sharing 
techniques, and bandwidth efficient Quadrature Phase Shift Keying (QPSK) 
modulation have become de facto standards for Layer 1 (physical layer) although 
1 0 many other techniques and modulations are used to some extent. 

Pure TDMA mesh networks exist, in which all user nodes take turns 
transmitting in a half duplex fashion. Pure FDMA point-to-point, or Single Channel 
Per Carrier (SCPC) networks, exist to allow full duplex transmission and reception. 
Many commercially viable network systems have evolved into a hybrid star topology, 
15 as shown in Figure 1, taking advantage of the broadcast nature of geo-synchronous 
satellites by using a SCPC downstream carrier from a central hub to all user nodes, 
and using one or more TDMA upstream(s) shared by all user nodes to communicate 
with the hub. 

Figure 1 illustrates a conventional sateUite communication network. A 
20 plurality of user nodes 12 communicate with a hub 1 1 via a satellite 10. The hub 1 1 
transmits data to the user nodes 12 via broadcast downstream channels 13. Each of 
the user nodes 12 transmit data to the hub 1 1 via upstream channel 14. 

A very popular and simple implementation of the TDMA upstream channel is 
to use an ALOHA technique enabling any user node to transmit to the hub any time 
25 the user node has data to send and relying on the probability that no other user node 



2 



chose to transmit at that time, which would cause a coUision. Once the demand for 
bandwidth exceeds 18.4% of the upstream channel (or 36.8% for slotted ALOHA), a 
more complex Layer 2 (media access control layer) methodology is required to 
schedule TDMA bursts so that collisions are prevented. 

Many such Layer 2 methods exist, ranging from static configurations that 
permanently assign a given time slot to each user node, to more dynamic 
configurations which schedule bursts to service specific needs of individual user 
nodes. 

As with any network system, the usefulness of control techniques at Layer 2 or 
higher is dependent on reliable performance of Layer L The geo-stationary satellite 
channel is well known to have an extremely long one-way delay (up to -300ms) and it 
is extremely power limited due to the long propagation delay which results in the need 
to operate at the lowest signal-to-noise-ratio (SNR) possible in order to minimize the 
cost of radio-frequency (RF) equipment. Additionally, the channel is subject to 
further attenuation due to atmospheric conditions, some of which are variable, such as 
rain fade. If the SNR becomes too low, the bit error rate increases to an unacceptable 
level and all communication breaks down and creates adverse effects up the entire 
protocol stack. In particular, the Layer 2 slot allocation process breaks down very 
quickly for reliable protocol-protected data since the bandwidth demanded by user 
nodes is allocated by the system, but the user nodes' demand level is not reduced 
since data is lost and must be retransmitted. The situation worsens when the SNR is 
reduced so badly due to a fade condition that even station-keeping control information 
cannot get through and the user node actually drops out of the network. In this case 
the slot allocation process may waste valuable bandwidth before learning of the 
outage by allocating slots to the user node that the user node is imable to use. 



Problems continue to arise as modem higher-level protocols are considered 
such as Voice over IP (VoIP) and streaming media which are sensitive to packet jitter. 
These protocols are designed to overcome the jitter of Internet Protocol (IP) traveling 
over terrestrial connections where the jitter is primarily due to queuing delays; 
however, the jitter introduced by a shared TDMA upstream satellite channel is often 
enough to degrade performance even when quality of the Layer 1 connection is 
excellent. 

Any Layer 2 control protocol designed to maximize throughput and mitigate 
the performance problems experienced by users of satellite networks is itself subject 
to the same problems as the application data packets. Traditional control methods 
requiring handshaking are not very helpful since it takes too long, for example, to 
command one user node to stop transmitting on a TDMA slot, wait for an 
acknowledgement, and then command another user node to begin. Adding multiple 
upstream channels such that user nodes may transmit on any one of several channels 
helps matters somewhat since it increases the possible number of transmission 
opportunities per frame, however it does not solve the problem of the delay incurred 
while changes are made to the slot allocation. 

An additional Layer 1 problem in traditional satellite networks, as recognized 
by the present inventors, is that the network's link budgets are over-designed due to 
the lack of adaptability of the modulation, demodulation, and FEC processes used 
therein. For example, if the threshold SNR of a link to achieve the minimum BER is 
5dB, a conventional link budget will include the amount of power required to achieve 
a 5dB SNR plus XdB of rain margin (see Pritchard). The conventional approach is 
based upon the assumption that when it rains, there will be up to XdB of atmospheric 
attenuation so an SNR of (5+X)dB is established during clear sky conditions to ensure 



an SNR of at least 5dB during heavy rain. During clear sky conditions, which occur 
most of the time for much of the world, XdB of link margin are wasted in the 
conventional approach. 

Figure 2 illustrates additional problems with conventional network 
communication methods, as recognized by the present inventors. This figure 
represents two timing scenarios, labeled as follows: No Synchronization, and 
Conventional Synchronization. In each scenario, HRxSOFn(ta) represents the 
nominal time a start of a data burst time frame (i.e. Start of Frame (SOF)) is received 
from user node "n," and kniax*2 represents the maximum variation of reception time, 
or tracking error, for the worst case user node, due to minor perturbations in satellite 
position. The frame's relative frame sequence is "a." 

In the No Synchronization scenario, SOFs are unconstrained and arrive at 
various times due to variations between the propagation delays of each user node. In 
the Conventional Synchronization scenario, a time Rcdn (Remote Conventional Delay 
for User Node n) is added at each user node so that SOFs from all user nodes are 
received synchronously to the Hub's transmit reference SOF. 

The conventional approach synchronizes HRxSOFn of each user node, but 
arrival time variation resulting from satellite tracking error disadvantageously results 
in some HRxSOFn occurring during hub frame n and some HRxSOFn occurring 
during hub frame n-1. 

Further, in the conventional approach, Rcdn is chosen to synchronize 
HRxSOFn with subsequent Hub transmit SOF, which results in simultaneous arrival 
of SOFs sent at different times in the frame sequence. Thus, data bursts arriving at 
the Hub cannot be assumed to have all been sent during the same frame time, thereby 



complicating network control methods and increasing response time to control 
commands sent from the Hub. 

As recognized by the present inventors, the industry is in need of a network 
system that combats these problems and enables standard off-the-shelf networking 
equipment using standard protocols to take advantage of a communication satellite's 
ability to reach user nodes at extreme distances and in areas of the world where 
broadband terrestrial communication is otherwise not practical. 

SUMMARY OF THE INVENTION 

Accordingly, one object of this invention is to address the above-identified and 
other problems with conventional systems and methods. 

The present invention provides a method to enable a satellite network system 
to dynamically monitor and adapt in real time to changing conditions on each satellite 
channel and changing traffic patterns at the user nodes. Another object of this 
invention is to maximize the quaUty of service that all users of the network experience 
by ensuring that each user node is communicating with the hub station at a higher data 
rate than conventionally possible while staying above a threshold bit error rate, and 
fiirthermore, to ensure that the TDMA upstream channel is shared amongst all users 
and applications according to predefined rules. 

At least three processes and their associated signaling protocols are used to 
accomplish this goal. The Demand Assignment (DA) process makes a decision every 
frame (40ms + processing or greater) about how many TDMA time slots out of an 
available pool of time slots will be allocated to each user node to transmit data to the 
hub station. An Uplink Control (UC) process constantly (or at least nearly constantly) 
monitors channel assessments made by the hub receiver every time a packet is 



received, sorts the assessments by user node, and makes a decision as to whether a 
user node should modify its frequency offset, symbol timing offset, transmit power, 
modulation, or FEC code in order to maintain channel quality as good or better than a 
threshold bit error rate (BER). Finally, a Downstream Recovery (DR) process 
monitors the remote status messages received periodically from each user node and 
determines from the dovrastream SNR and number of cyclical redundancy check 
(CRC) errors reported whether the node has fallen below the threshold BER. In this 
case, or the case that communication is lost completely with a user node, the DR 
process begins to send recovery packets to the node using an alternate, more robust, 
modulation (e.g., Bi-Phase Shift Keying (BPSK) modulation) using more energy per 
bit and/or a more powerful FEC code to try and reestablish communication. 

To enable the DA and UC processes to work reliably and react quickly to 
changing channel conditions, a synchronous Burst Time Plan (sBTP) is employed. 
The sBTP ensures that the propagation delay is a same integer number of frames for 
all node-to-hub paths. This enables time slots, modulation, and FEC to be changed 
every frame even though that frame is much shorter than the propagation delay 
between the hub and the user nodes. 

Since rain fade is a physical property of the satellite charmel, it is not possible 
to directly counteract its effect without making a tradeoff. The traditional tradeoffs 
include either over-designing the link (which requires using more expensive RF 
equipment and possibly wasting bandwidth by power limiting the link), or allowing 
the BER to increase (which could become so bad that it causes a complete outage). 
The UC and DR processes take a different approach, which is to lower the effective 
data rate of the link by changing modulation or FEC during the rain fade without BER 
degradation. Using traditional methods, the link is designed for some availability 



based on the percentage of time during the year that there is heavy rain at the user 
node location. For example, if it is expected that there will be rain fade of greater 
than 3dB 0.01% of the year, the link can be over-designed by 3dB to produce a link 
that is 99.99% available. If the same link were designed using a UC and DR process, 
the link could be designed with no rain margin. When a rain fade occurs, UC and DR 
sense the condition and change to an alternate modulation and FEC code which 
requires 3dB less SNR in order to maintain the desired BER. The consequence of 
using the alternate modulation and FEC is that the data rate of the link is reduced (e.g., 
switching from QPSK to BPSK halves the number of bits per second, and increasing 
the FEC rate could reduce the data rate further). However, as recognized by the 
present inventors it is preferable to avoid the cost of over-designing the link in 
exchange for operating at a lower data rate during the infrequent periods of heavy rain. 
In fact, the UC and DR capability results in users getting higher performance than 
they would using a traditional link design by choosing not to decrease the cost of RF 
equipment, but instead to use the rain margin that is no longer required to increase the 
bit rate that they can operate under clear sky conditions. Either way, UC and DR 
allow the users a valuable choice that they would not otherwise have. 

BRIEF DESCRIPTION OF THE DRAWINGS 
A more complete appreciation of the invention and many of the attendant 
advantages thereof will be readily obtained as the same becomes better understood by 
reference to the following detailed description when considered in connection with 
the accompanying drawings, wherein: 

Figure 1 illustrates a conventional communication network system; 

Figure 2 is timing diagram of a conventional commimication network system; 



8 



Figure 3 illustrates a communication network system according to the present 
invention; 

Figure 4 is a block diagram of a hub according to an embodiment of the 
invention; 

5 Figure 5 is a timing diagram of a communication network system according to 

the present invention; 

Figure 6 is an example of Rfsd and Hfsd timing diagram according to the 
present invention; 

Figure 7 is an example timing diagram of synchronous BTP according to the 
10 present invention; 

Figure 8 is a flow chart of a communication network control method according 
to the present invention; 

Figure 9 is a failsafe method state diagram according to the present invention; 
Figure 10 is a demand assignment signal flow diagram in a communication 
15 network control method according to the present invention; 

Figure 1 1 A is a queuing diagram for a communication network control 
method according to the present invention; 

Figure 1 IB is a diagram of a demand header calculation process in a 
communication network control method according to the present invention; 
20 Figure 12 is a communication network control method signal flow diagram 

according to the present invention; 

Figure 13 is a chaimel group diagram for a communication network control 
method according to the present invention; 

Figure 14 is a slot allocation process flow diagram for a commxmication 
25 network control method according to the present invention; 



9 



Figure 15 is an example diagram of a feathering process for a communication 
network control method according to the present invention; 

Figure 16 is an example diagram of a feathering process across multiple 
channels in a channel group according to the present invention; 

Figure 17 is an example diagram of a feathering process to minimize channel 
hopping in a channel group according to present the invention; 

Figure 1 8 is an upstream control process flow diagram in a communication 
network control method according to the present invention; 

Figure 19A is a downstream recovery process flow diagram in a 
communication network control method according to the present invention; 

Figure 19B is a QPSK constellation used in the present invention; 

Figure 19C is a BPSK constellation used in the present invention; 

Figure 20 is a flow diagram of a downstream recovery process in a 
communication network control method according to the present invention; 

Figure 21 is a block diagram of an exemplary processor apparatus according to 
the present invention; and 

Figure 22 is a block diagram of an exemplary user node according to the 
present invention. 

DESCRIPTION OF THE PREFERRED EMBODIMENTS 
Referring now to the drawings, wherein hke reference numerals designate 
identical or corresponding parts throughout the several views, and more particularly to 
Figure 3 thereof, which illustrates an embodiment of a satellite commimication 
network according to the invention. A plurality of user nodes 304 communicate with 
a hub 302 via a satellite 300. The hub 302 transmits a sequence of frames to the user 



10 



nodes 304 via broadcast downstream channel(s) 308. Each of the user nodes 304 
transmit bursts of data to the hub 302 via shared TDMA upstream channel(s) 310. 
The hub 302 is further connected (via a wired or wireless connection) to a LAN/WAN 
314 and thereby communicates with a plurality of devices including: network 
operator access devices 318 used to control system parameters; data service devices 
320 including web servers, email servers, web/DNS caching servers, video archive 
servers, and other related devices; voice service devices 322 including VoIP gateways 
and other devices connected to the public switched telephone network; and intemet 
service devices 324 including routers and other connection points to the global 
intemet. Each of the user nodes 304 is further connected to a LANAVAN 3 14, and 
thereby communicates with a plurality of client computers 316 and client peripherals 
326, including video conferencing devices, web servers, LAN routers with 
connections to other networks, VoIP routers with connections to telephone and fax 
devices, and other peripherals. An embodiment of the hub 302, illustrated in Figure 4, 
includes a central control server 400 (a more detailed example of which is shown in 
Figure 21), connected via a bus 402, to a hub transmitter 404 and a plurality of hub 
receivers 406. 

For example, an embodiment of the present invention may be used by a 
business with remotely located organizations or users that otherwise have no 
conventional means of accessing the global intemet or an intranet. Client computers 
316 connected to a user node 304 could include a single PC or an entire LANAVAN 
with many PCs, VoIP equipment, videoconferencing equipment, web servers, or any 
other piece of equipment that can communicate directly or indirectly using IP packets 
over Ethernet. Network operators may, for example, use the present invention to 
provide satellite connectivity to these users. The network operator is responsible for 



11 



administering at least the satellite portion of the network. The operator has special 
access to the central control server 400, which configures, monitors, and controls the 
network. Network control tasks may include adding and removing user nodes 304, 
and changing data rate and QoS parameters, etc. . . An example of a hub installation 
according to the present invention includes a router connected to the central control 
server 400 and, via the LANAVAN 314 and internet service devices 324, connected to 
the global internet. This example configuration allows client computers 316 to access 
the intemet over the satellite link by designating the IP default gateway of each of the 
client computers 3 16 as the user node's IP address. This causes any packets 
generated at the user node's LANAVAN 314 destined for IP addresses that are not 
local to be forwarded to the satellite router and over the air to the hub 302, Similarly, 
the default gateway of the central control server 400 is set for the router connected to 
the global intemet. This causes packets that arrive at the hub 302 with IP addresses 
not matching any of the subnets on the satellite network to be forwarded to the router 
and then on to the global intemet. 

In addition to a global intemet connection, the network operator may have 
value added services running at the hub's LANAVAN. An example embodiment of 
the present invention includes a VoIP gateway voice services device 322 which is 
connected to the LANAVAN 314 at the hub and also to a Public Switched Telephone 
Network (PSTN). In this example, the user nodes 304 use VoIP equipment to call 
phone numbers on the PSTN using the sateUite network as a transport. Also in this 
example, web and email server and caching equipment intemet service devices 324 
connected to the hub 302 offer special services to the clients connected to user nodes 
304. 



12 



In addition, although only a communication network including a geostationary 
satellite has been discussed, the present invention also applies to communication 
networks without a geostationary satellite, but instead incorporating other physical 
layer features. For example, the network may include optical transceivers, wireless or 
WiFi network devices, microwave transceivers, broad beam or other low gain antenna 
based devices, low-earth, medium-earth, or elliptical orbiting satellites, and other 
physical layer links as hubs and/or user nodes, when a delay is involved, either due to 
propagation, or perhaps due to latency (e.g., a WiFi based hub that ' Vakes-up" at 
predetermined intervals to transmit data from/to user nodes, which may be other WiFi 
devices). Furthermore, terrestrial-based RF-microwave transceivers may be used as 
well. 

Synchronous BTP 

To synchronize user nodes with a hub according to one embodiment of the 
invention, a round trip delay between each hub/user node pair is set equal to a 
common integer number of frame lengths. A duration of the selected frame length 
influences system performance, as described below. In addition, all user nodes 
receive the Burst Time Plan (BTP) and apply it in a manner so that no two user nodes 
collide with one another when data arrives at the satellite (or hub receiver). These 
two requirements are satisfied by generating a system timing reference at the hub 
transmitter (hub TxSOF) and applying offsets at the hub receivers (Hfsd) and user 
nodes (Rfsd) to this reference. Each user node has its own Rfsd based on its geographic 
location. In order to provide the maxiinum amount of time for the user nodes to 
process a BTP, these offsets are applied optimally. 

Frame and Symbol Synchronization 



13 



The hub and all user nodes are referenced to a Transmit Start of Frame Sync 
Marker (TxSOF). This frame sync is generated by the hub transmitter periodically at a 
fixed rate. The time of arrival at each node of the frame sync is calculated based on 
the line of slight distance traveled by each signal between the hub, satellite and user 
nodes, and the known speed of radio signals along this path. The arrival time of each 
burst at satellite, and in tum at the hub receiver, is controlled by setting an appropriate 
fixed delay at each user node (Rfsd), as described below, thereby achieving 
synchronization of each user node to within a symbol period. Frame synchronization 
in tum allows for BTP synchronization. Note that throughout this specification, the 
use of terminology such as "SOF transmitted by user node" or "user node SOF 
received at the hub" refer to the starting point of a virtual frame time period, that 
includes actual data bursts within that frame sent from one or more user nodes to the 
hub; although, the user nodes do not send an actual SOF signal. SOFs from the user 
nodes should be thought of as the start of a virtual frame time period. 

Figure 5 depicts the timeline for an exemplary embodiment of synchronized 
BTP in an example network consisting of a hub and three user nodes. One user node, 
user node #1, is located closest to the satellite, user node #3 is located fiirthest from 
the satellite, and user node #2 lies in between (in distance/time) of user nodes # 1 and 
#3. In this example, a BTP is first transmitted from the hub to the satellite with a 
delay indicated as th-s. The BTP is then transmitted from the sateUite to each of the 
three user nodes. The propagation delay of the BTP from the satellite to the user 
nodes is depicted by the three dotted lines extending from sateUite TxSOF to Rxl 
SOF, Rx2 SOF, and Rx3 SOF, and the propagation delays of the signals are ts.ri,ts-r2, 
and ts-r3, respectively. A unique delay, Rfsdn, is applied at each node before a start of 
frame time is transmitted to the satellite. From Figure 5, it can be seen that the 



14 



shortest Rfsd is applied at user node 3, which is the greatest distance from the satellite, 
and the longest Rfsd is applied at user node 1 which is the user node that is closest to 
the satellite. After applying the respective delay Rfsd at each user node, each user 
node transmits data bursts in a frame time period to the satellite with start of frame 
time as depicted by the dotted lines going from Txl SOF, Tx2 SOF, and Tx3 SOF to 
the satellite Rx SOF. Further, as can be seen in Figure 5, each Rfsd value for a 
particular user node is selected such that all start of frame times transmitted by the 
user nodes arrive at the satellite simultaneously. Subsequently, the satellite transmits 
the received data bursts from the user node to the hub with a propagation delay of th-s- 
As shown in Figure 5, the arrival time of the start of frame time periods from each 
user node is offset from a previous start of time frame to be sent from the hub by 
HUBfsd. 

Figure 6 illustrated the roimd trip delay in the signal path from a hub Tx SOF 
to a hub Rx SOF, including delay Rfsd for a particular user node. In this example, the 
propagation delay from hub Tx SOF to satellite Tx SOF is indicated as th-s, the 
propagation delay from satellite Tx SOF to remote Rx SOF is indicated as ts-r the 
delay time applied at the user node tfsd is applied between the time of remote Rx SOF 
and remote Tx SOF at the user node. The propagation delay time from remote Tx 
SOF until satellite Rx SOF is illustrated as ts-r, and the propagation delay from 
satellite Rx SOF until hub Rx SOF is indicated as th-s. Further, hub Rx SOF is 
received Hfsd after the previous hub Tx SOF in frame 6. 

Calculation of the Rfsd for each user node depends upon a Hub Frame Start 
Delay, Hfsd. The Hfsd is based on: 



15 



• location of a virtual user node (vr) at a location on the earth which is closest to 
the satellite, a known constant, which determines a signal propagation delay 
between the satellite and the virtual user node, tsvr 

• location of the satellite relative to the hub, which determines a signal 
propagation delay between the satellite and the hub, ths 

• ground delay associated with latencies in the hardware on both the hub and the 
user node, tgd 

• frame rate, F 

• maximum frame start delay, FSDmax = 1/F - k, where k is a constant that 
accommodates satellite movement from its "center" location to its location 
when it is closest to earth. 

Hfsd = mod [(2ths + 2tsvr + tgd + FSD^ax), 1/F] 

The Rfsd calculation for each user node is based on: 

• hub frame start delay, Hfsd 

• location of the user node relative to the satellite, which determines a signal 
propagation delay between the satellite and the user node, tsm 

• location of the hub relative to the sateUite, which determines a signal 
propagation delay between the hub and the satellite, ths 

• frame rate, F 

• ground delay associated with latencies in the hardware on both the hub and the 
user node, tgd 



16 



Rfsd = 1/F - mod[(2ths + 2tsni + tgd - Hfsd), 1/F] 



Calculation and Distribution of Rfsd 

The Hfsd is calculated during initialization on the hub receiver using pre- 
configured hub and satellite location, and frame rate. Each user node is pre- 
configured with its location, or determines its location from a GPS input. Frame rate, 
and other user node parameters, including upstream symbol rate, are sent to each user 
node via the downstream channel and stored at the respective user node before the 
user node acquires into the network. When each user node is instructed to acquire 
into the network, it is also informed of the Hfsd (or of the variables needed to calculate 
it locally), the location of the satellite, and the location of the hub. The user node 
calculates a Rfsd based on this information, prior to transmission. This method 
accommodates both static and mobile user nodes, and the movement of the satellite - 
since the user node recalculates the Rfsd each time the user node is instructed to 
acquire into the network. 

As stated earlier, the hub transmitter uses the hub transmitter's TxSOF to start 
transmitting a new downstream frame in the modulator. At each user node, this 
downstream frame boundary is delimited and a receive start of frame RxSOF is 
generated. The RxSOF on each user node is delayed by a programmable frame start 
delay Rfsd, calculated as above. With the Rfsd applied at each user node, the start of all 
upstream frame times arrive simultaneously (i.e., within the ability of the hub 
demodulator to demodulate the burst within the same symbol time) at the satellite, and 
hence simultaneously at the hub receiver. Individual bursts within the frame are 
controlled by the BTP, see below. At the hub receiver, each upstream arrives at a 
fixed offset, Hfsd relative to the hub transmitter's TxSOF, so that the generated RxSOF 



17 



time period coincides with the start of the upstream frame. This allows the bxirsts to be 
extracted from the demodulated signal. 

To illustrate an embodiment of the invention with respect to a conventional 
approach, Figures 2 and 5 illustrate the following three timing scenarios: No 
Synchronization (Figure 2), Conventional Synchronization (Figure 2), and 
Synchronous BTP Timing (Figure 5). In each scenario, HRxSOFn(ta) represents the 
nominal time a start of a frame time period. Start of Frame (SOF), is received from 
user node "n," and kmax*2 represents the maximum variation of reception time, for the 
worst case user node, due to minor perturbations in satellite position. The frame *s 
relative frame sequence is "a." 

In the No Synchronization scenario, SOFs are unconstrained and arrive at 
various times due to variations between the propagation delays of each user node. In 
this example, SOFs from user nodes 1 and 2 transmitted at frame sequence tl, 
HRxSOFl(tl) and HRxSOF2(tl), respectively, are received at the hub prior to the hub 
TxSOF for frame n, while the SOF from user node 3 transmitted at frame sequence tl, 
HRxSOF3(tl), is received at the hub after the hub TxSOF for frame n. In the 
Conventional Sync scenario, a delay time Rcdn (Remote Conventional Delay for user 
node n) is added at each user node so that the SOFs from each user node are received 
synchronously, coincident with a hub TxSOF. The diagram illustrates two problems 
with this conventional approach: 

1 . Although the conventional approach nominally synchronizes 

HRxSOFn of each user node, each particular user node is suffering 
under a different tracking error at any given time. Thus, tracking error 
may resuU in some of the HRxSOFn being received during hub frame 
n and some HRxSOFn being received during hub frame n-1 . In this 



18 



example, HRxSOF2 is received less than half of the total variation of 
reception time due to satellite perturbations, or kmax, before the hub 
TxSOF, thus, depending upon the current location of the satellite, 
HRxSOF2 is sometimes received during frame n-1 and synchronized 
to the Hub TxSOF for frame n and sometimes received during frame n 
and therefore synchronized with hub TxSOF for frame n+1, resulting 
in unpredictable timing. 
2. The Conventional Synchronization approach results in simultaneous 
arrival of SOFs sent at different times in the frame sequence. In this 
example, HRxSOFl(tl) and HRxSOF2(tl) both transmitted at frame 
sequence tl, are synchronized with HRxSOF3(tO), which was 
transmitted during the previous time period, tO as shown in Figure 2. 
Figure 7 illustrates signal timing at a hub with synchronous BTP according to 
an embodiment of the invention. HRxSOFl, HRxSOF2, and HRxSOFS, are each 
received simultaneously at the hub. The time of arrival of these HRxSOF signals is 
Hfsd after each hub TxSOF. Further, the HRxSOFl, HRxSOF2, and HRxSOFS 
received simultaneously, were all transmitted relative to the same frame sequence 
from their respective user nodes, kmax *2 indicates the potential variation in time of 
receipt of each HRxSOF signal due to variation in satellite position, and Tmargin 
represents the amount of time from the preceding TxSOF until the earhest potential 
arrival time of the HRxSOF signals. 

The timing in Figure 7 illustrates an exemplary solution to both problems 
described above. First, a time Hfsd is added to each user node to shift the arrival times 
of all SOFs closer to the middle of a hub frame time, such that Tmargin is maximized, 
and so the user node closest to the satellite has the maximum Rfsd. Second, Rfsd is 



19 



selected for each user node (based on Hfsd) such that SOFs are received 
simultaneously with respect to frame sequence. Thus, in this example, all SOFs 
received at HRxSOFl-3 in Figure 7 were transmitted by user nodes at the same frame 
time in the sequence of frame times transmitted by each user node. 

A communication network control method according to an embodiment of the 
invention is further illustrated in Figure 8. The process begins in step S800, where the 
hub delay is calculated. The process then processes to step S802, where the hub delay 
and other parameters are transmitted to each user node. The process continues in 
S804, where a transmission delay for each user node is calculated using the received 
parameters and received Hfsd. Next, in step S806, the hub transmits a sequence of 
frames to each of the user nodes. Further, the process continues in step S808, where 
each burst is delayed by Rfsd, and each of the user nodes transmits data bursts in a 
frame time period to the hub, in step S810. Finally, in step S812, the frame time, 
including data bursts from each user node, are received simultaneously at the hub, Hfsd 
after the start of a hub frame time. Thus, the Rfsd for each node is selected to ensure 
that data burst frame times from each user node arrive at the satellite simultaneously, 
and are then transmitted to the hub, where they are also received in their respective 
slots according to the frame timing. * 

BTP Synchronization 

Frame synchronization as detailed above provides the ability for bursts within 
a frame to be optimally used without the fear of a burst from one user node colliding 
with the burst of another user node on the same upstream. The synchronous 
broadcasting of a BTP from the hub to all user nodes is used to control which 



20 



upstream channel, and which bursts within that upstream channel, are enabled for a 
specific user node. 

The hub transmitter uses the hub transmitter TxSOF to: 

generate an interrupt to the BTP process a configurable time period 
prior to the TxSOF. The interrupt is earlier than TxSOF to account for all hardware 
latencies in both the hub transmitter and user node receivers. This allows the BTP to 
be placed in the downstream earlier than the actual start of frame and ensures that the 
user node is given the maximum amount of time to process and apply the BTP prior to 
the start of the user node's transmit start of frame; and 

• synchronize the start of the downstream frame encapsulation and FEC 
block boundaries. 

The interrupt to the BTP Process on the hub transmitter in tum is used to: 

select the next BTP from the BTP queues and broadcast it to the 
downstream user nodes; and 

trigger the BTP process to send a message to the central control server 
On detection of the BTP request message, the DA process on the central control 
server sends a new BTP to the hub transmitter. The hub transmitter inspects packets 
coming from the central control server and transmits those containing BTPs ahead of 
other enqueued packets. 

On each user node, the BTP is extracted from the downstream frame by the 
BTP process and applied to the burst controller immediately. At the appropriate time, 
the burst controller enables the modulator to transmit data onto the selected upstream 



21 



carrier. The BTP determines both the frequency of the burst and in which slot or slots 
in the frame the data is transmitted. 

A failsafe mechanism prevents errant BTP from being applied to the burst 
controller. 

Figure 9 illustrates the sequence of events in an exemplary embodiment of 
BTP synchronization by the burst controller each frame. First, the TxSOF pulse 
occurs, bringing the process to step S902, where it waits for a BTP update. Then the 
BTP memory is updated with the next frame's information, which brings the process 
to step S904 where it waits for the BTP to be armed. Finally the BTP is armed 
bringing the process to step S906 which indicates that data will be sent when the 
current time slot number matches any that are allocated to the node for the current 
frame. In the event that the correct sequence is not followed, the process is brought 
back to step S900, and the BTP is ignored until a valid sequence has been received 
and no further bursts are transmitted in the interim. A failsafe mechanism is needed 
because it is possible , due to factors such as excessive processor loading, that an 
incorrect sequence may occur that could apply an incorrect BTP and cause collisions. 

A BTP process is also running at each hub receiver and is applied to a burst 
controller. Since there is an integer number of frame times "n" from the time a burst 
is transmitted at the hub to the time when a retum burst arrives at the hub receiver 
(taking into consideration Hfsd), synchronization is advantageously simplified. The 
hub buffers "n" BTPs and applies the BTP that was sent "n" frames previously. The 
burst controller at the hub receiver informs the demodulator as to which modulation 
and FEC to use since it knows the current settings of the node that sent each burst. 
Demand Assignment Process 



22 



The DA process ensures that TDMA slots on the upstream channel are 
allocated in a manner consistent with the resource sharing rules specified by the 
network operator or user. These rules are applied on a node-by-node basis and 
involve rate shaping and Quality of Service (QoS) priorities. 

Rate shaping rules govern the data rate allocated to a user node, such as a 
minimum, maximum, or Committed Information Rate (CIR = guaranteed when 
required). A benefit of this process is that there is no requirement that this data rate 
maps evenly to a number of TDMA slots, and therefore the network operator defining 
the rate shaping rules may be agnostic to the TDMA implementation. 

QoS priority rules govern the relative priority of different types of data that 
may arrive at the user node and destined for the hub or a destination on a network 
(such as the internet) connected to the hub. It is quite common for several user nodes 
to have identical rules. For example, the network operator may choose to limit all 
nodes to a maximum of 100kbps. When allocating slots to user nodes with identical 
rules, the DA divides the slots equally. 

Figure 10 illustrates an exemplary embodiment of the Demand Assignment 
Process. An interrupt SI 000 occurs at a programmable interval triggered by a fi-ame 
marker in the hub transmitter. A Burst Time Plan (BTP) generator assigns TDMA 
slots based on data structure and upstream parameters in the central control server, in 
step SI 002. BTP is multiplexed into a downstream broadcast channel in the hub 
transmitter, in step 81 004. BTP is decoded by each node, and intemal buffering is 
modified based on the number of slots to be transmitted in the upcoming fi-ame in step 
SI 006. Each node updates its transmitter to schedule bursts on the TDMA slots it has 
been allocated on the upcoming fi-ame in step SI 008, and synchronous BTP firmware 
ensures that data is transmitted during scheduled slots with an updated Demand 



23 



Header added in SI 010. In the hub receiver, TDMA bursts are received and error 
checked. Good (i.e., not detected as being corrupted) packets are forwarded to the 
central control server in step S1012. Li the central control server, demand data is 
sorted, in step SI 01 4, by user node into its corresponding position in a Demand Data 
Structure 1050. 
DA at the User Nodes 

Components of the DA Process exist at every user node where demand 
information is collected and packets are queued according to the current allocation of 
TDMA slots. Every single packet that is transmitted over the upstream channel is 
prepended with a header to reflect the current demand levels of the user node at the 
time when that packet was sent. This ensures that the DA decisions made by the 
central control server are using the most current information possible. 

Figure 1 1 A illustrates an exemplary embodiment of a portion of the DA 
process at the user nodes, hi this example, the process uses a queuing system that 
separates data into different queues according to QoS service levels, a demand 
calculator, and an adjustable length transmit queue. Each IP packet 1 100 arriving 
from a user node LAN is classified according to the QoS Service Level Rules 
specified by the network operator in step SI 102, and then segmented into blocks the 
size of the TDMA payload in step SI 104. Segments 1 108 are distributed into 
different queues 1110, each corresponding to a different QoS Service Level, and then 
segments are taken from Service Level queues by a weighted fair queuing step SI 1 12, 
appended with DA Headers in step SI 1 14 and placed into the Transmit Queue 1118 
based on the weighted priority of each QoS Service Level. Latency is minimized by 
adjusting the length of the Transmit Queue 1 1 18, in step SI 1 16, in proportion to the 



24 



number of TDMA slots allocated to the user node on the upcoming frame, based upon 
the BTP updated via downstream update on every frame. 

The queuing system uses service levels defined by a set of rules describing 
properties of an incoming packet. A simple QoS configuration may designate one 
service level for TCP packets, another for UDP packets, and another for everything 
else. More complex service levels are defined according to IP address ranges, TCP 
port numbers, or any other parameters that are used to classify packets. It is important 
to note that the data units enqueued in these service level queues are not complete IP 
packets, rather, they are segmented packets divided up such that each segment fits 
into the payload portion of a TDMA burst. By segmenting, finer granularity is 
achieved so the weighted fair queuing decisions are made on a burst-by-burst basis 
using identical sized units, rather than on a packet-by-packet basis using variable 
sized units. The resuh of using this segmentation is a lower latency, optimally fair 
queuing system because it avoids the situation where a large packet monopolizes the 
transmit queue while other packets (potentially of higher priority) build up in the 
service level queues. 

Each service level queue has two associated "flags" to identify special 
properties of the queue: the Committed Information Rate (CIR) Trigger flag, and the 
Real Time flag. The flags are either checked or unchecked by the network operator 
when the QoS rules for the service level are specified. If the CIR Trigger flag is 
checked for a given service level, the queue is authorized to request that the user node 
be granted bandwidth, up to the CIR specified by the network operator in the rate 
shaping rules, for as long as segments are pending in that queue. Similarly, if the 
Real Time flag is checked for a given service level, the segments in that queue are 
given Real Time status when calculating the total demand of the user node. CIR 



25 



Trigger status and Real Time status are mutually inclusive because a network operator 
may, for example, want to designate VoIP traffic as Real Time and also allow that 
traffic to request the CIR specified for that node. 

The demand calculator is essentially a compression method in that it collects 
complex information about the demand of the user nodes and compresses it into a 
very small header summarizing the information. Communicating the node's demand 
concisely is important for two reasons. First, bandwidth on the upstream chaimel is a 
precious commodity due to the cost of the satellite segment and the fact that the 
channel is shared by many users. Wasting too much bandwidth on demand request 
overhead diminishes the overall benefit of a demand driven system. Second, limiting 
the number of input parameters to the DA process on the central control server serves 
to simplify its implementation, thereby increasing the speed with which decisions may 
be made. 

The three fields in the demand header are Total Demand, % Real Time 
Demand, and CIR Trigger. Total demand is equal to the total number of TDMA slots 
required to satisfy the demand at the user node. An additional overhead factor may be 
used to increase the Total Demand number to account for the overhead data, such as 
the demand headers themselves, which are sent with the segments. The % Real Time 
Demand is a number that represents the percentage of the Total Demand generated by 
service level queues flagged as Real Time. The CIR Trigger is a flag in the header 
that is checked if any of the service level queues with CIR Trigger status contain 
pending segments. 

The adjustable length transmit queue at each user node optimizes latency and 
throughput of the upstream channel by avoiding two situations which could cause 
sub-optimal performance. The transmit queue "pulls" a segment from the service 



26 



level queues when the number of queued packets falls below the current queue length 
threshold. Consider a situation where five segments are in the transmit queue of a 
user node but 10 TDMA slots have been allocated to the node for the current frame. 
In this situation, the upstream transmitter may transmit all five segments before the 
transmit queue can "pull" five more and the result is that five transmission 
opportunities were missed and hence, bandwidth has been wasted. Consider a second 
situation where many low priority segments are enqueued in the transmit queue 
waiting to be sent while higher priority segments may have subsequently arrived at a 
high priority service level queue. In this case, the weighted fair queuing algorithm 
has already made a decision to put the low priority segments into the transmit queue 
because it didn't know that a high priority segment was going to arrive. Now the high 
priority segment waits in the service level queue until the transmit queue can 
accommodate it, hence the high priority segment's latency is increased and the 
relative priority wishes of the network operator have not been optimally honored. 

In theory, a solution to this problem would be to have no transmit queue at all 
and to take packets directly from the service level queues and feed them directly into 
the upstream transmitter. In practice, however, it is very difficult to synchronize the 
timing of the frames (which must be accurate to the resolution of one modulation 
symbol time) with the user node's control processor (whose response time varies with 
loading). Therefore, a good solution is to adjust the length of the transmit queue in 
direct proportion to the number of TDMA slots (transmission opportunities) that have 
been allocated to that node for the current frame. The proportionality constant 
goveming the relationship between allocated slots is implementation dependent. In 
the case where the frame is fairly long and the control processor on the user node is 
fairly fast, a proportionality constant less than one is chosen to yield a transmit queue 



27 



shorter in length than the number of slots allocated. Conversely, if the upstream 
frame is short and the processor is relatively slow, a proportionality constant greater 
than one is chosen to avoid wasting transmission opportunities. 

Figure 1 IB further illustrates an exemplary embodiment of the present 
invention including the collection of parameters from each Quality of Service Queue 
1110. The collected parameters include a CIR Trigger 1 126, Real Time 1 120, Arrival 
Rate Measurement 1 122, and Depth Measurement 1 124. In this example, these 
collected parameters and the configuration of the associated queue are used by a 
calculating demand step SI 1 14 to calculate new demand header 1 136 values every 
time a segment is transmitted. The demand header 1 136 includes a total demand field 
1 130, which represents the total number of slots required to satisfy the demand, a % 
real time field 11 32, which represents the percentage of the total demand generated by 
queues marked as Real Time, and a CIR trigger field 1 134, which indicates whether 
any of the demand is generated by queues that are entitled to request that the user 
node's CIR be allocated. 

Figure 12 illustrates upstream packet flow in an exemplary embodiment of the 
invention. In a user node in this example, a demand header 1202 is added to each 
TDMA packet or payload 1200. In the hub receiver, an assessment trailer 1204 is 
added to each payload 1200. In the central control server, the assessment trailer 1204 
and demand header 1202 are stripped from the payload and sent to the Upstream 
Control Process and the Demand Assignment Process, respectively. 
DA at the Central Control Server 

A decision-making component of the DA process resides at the central control 
server. Inputs to the DA decision module are real time estimates of each user node's 
demand, configuration parameters for the upstream channel group consisting of one or 



28 



more upstream chamiels, rate shaping rules defined by the network operator for each 
user node, and high level priority directives governing the relative priority of Real 
Time, CIR, and statically assigned TDMA slots. 

Figure 13 illustrates an embodiment of an Upstream Channel Group 1300 with 
M Channels 1306, each on a separate frequency, with N (or a multiple of N) time slots 
1304 repeating every Frame 1308. 

The demand assignment decision process is started every frame by the hub 
transmitter by sending a BTP request packet to the central control server shortly 
before a new fi-ame starts. When the decision process is kicked off, a demand-driven 
slot allocation decision is made to dole out a certain number of slots to each user node. 
The list of user nodes and the number of slots each has been awarded is passed to the 
feathering algorithm to optimally spread the allocated slots over the fi-ame so as to 
minimize the amount of packet jitter that will be introduced by the system. Finally, 
and without changing the feathering that was just performed, the slots assignments are 
divided up into separate upstream channels and placed into a BTP packet which is 
sent to the user nodes via the downstream channel. An additional step modifies the 
BTP in order to minimize the number of times each user node will have to hop in 
firequency between upstream channels in order to use all the transmission 
opportunities it has been awarded. 

A detailed view of an exemplary embodiment of the DA is illustrated in 
Figure 14. A demand driven Slot Allocation Decision S1406 is made once per frame 
(i.e., each time a new BTP Request 1414 is received), and is based upon inputs from 
Upstream Channel Group Configuration Parameters 1400, Prioritization Rules 1402, 
Upstream Rate Shaping Rules 1404, Recent Frame State Information 1408, and 
Demand Data Structure 1050. Feathering S 1410 is performed to spread slot 



29 



allocations across the frame time period to minimize upstream jitter, and a new BTP 
Packet 1412 is sent every frame to the hub transmitter which multiplexes it into the 
downstream channel. 

For example, an embodiment of the present invention may include the 
following upstream channel configuration parameters (additional discussion of the 
parameters follows the example): 

• 100 slots per frame 

• two upstream channels each at 1 Mbps 

• 10 Frames per second 



In this example, the following prioritization rules are applied in this order: 

1 ) Minimum Data Rate: 

2) CIR 

3) Real Time Data 
Example Rate Shaping Rules: 



NODE 


Minimum Rate 


Maximum Rate 


CIR 


A 


10 kbps 


50kbps 


20kbps 


B 


20 kbps 


1000kbps 


0 


C 


10 kbps 


500kbps 


100kbps 


D 


50 kbps 


100kbps 


100kbps 


E 


10 kbps 


50kbps 


0 


F 


20 kbps 


50kbps 


30kbps 


G 


10 kbps 


1000kbps 


50kbps 


H 


50 kbps 


100kbps 


100kbps 



Example Real Time Demand Information (for current frame time) 



NODE 


Total Demand 


%Real Time 


CIR 


A 


0 slots 


0 


No 


B 


80 slots 


0 


No 


C 


40 slots 


25% 


No 


D 


10 slots 


0 


Yes 


E 


0 slots 


0 


No 


F 


5 slots 


0 


No 


G 


60 slots 


0 


Yes 


H 


40 slots 


50% 


No 



30 



Example Demand Assigmnent Decision Process Explanation 

• Two upstream carriers with 100 slots per frame each results in 200 slots 
available for the current frame. 

• 235 slots requested so DA will have to make a decision. 

• Per the Example Prioritization Rules, the Minimum Data Rate is satisfied first 
for all nodes regardless of their current demand. Since a single slot represents 
1 kbit of data, a minimum rate rule of 10 kbps is satisfied by allocating at least 
one slot per frame. 

• one slot contains 1000 data bits. 

Example slot allocation after satisfying the minimum data rate constraint: 



NODE 


Slots 


A 


1 slot 


B 


2 slots 


C 


1 slot 


D 


5 slots 


E 


1 slot 


F 


2 slots 


G 


1 slot 


H 


5 slots 



A total of 18 slots have been allocated in this example and 182 slots remain free. 

• Per the prioritization rules, CIR is satisfied next. 

• Node D is requesting 10 slots which equates to 100kbps. Node D's CIR flag 
is checked and is allowed a CIR of 100kbps. Node D has already been 
allocated 5 slots to satisfy it's minimum rate rule so it is allocated another 5 to 
satisfy its demand. 



31 



• Node G is requesting a total of 60 slots which equates to 600kbps. Node G's 
CIR flag is checked and its CIR is 50kbps. Node G already has 1 slot so it is 
given 4 more to satisfy its CIR. 

The example allocation after satisfying CIR is: 



NODE 


Slots 


A 


1 slot 


B 


2 slots 


C 


1 slot 


D 


10 slots 


E 


1 slot 


F 


2 slots 


G 


5 slots 


H 


5 slots 



A total of 27 slots have been allocated and 173 slots remain free. 

• Per the example prioritization rules. Real Time demand is satisfied next. 

• Node C is requesting 40 slots and 25% (or 10 slots) of its demand is Real 
Time. It akeady has one slot and it is allocated 9 more to satisfy all Real Time 
demand. 

• Node H is requesting 40 slots and 50% (or 20 slots) of its demand is Real 
Time. To completely satisfy Node H's Real Time demand, it would need 15 
more slots to increase its total to 20 slots. However since that would equate to 
a data rate of 200kbps, the maximum rate constraint of 100kbps would be 
violated. Therefore, Node H is allocated 5 more slots for a total of 10. 



The allocation after satisfying Real Time demand: 



NODE 


Slots 


A 


1 slot 


B 


2 slots 


C 


10 slots 


D 


10 slots 



32 



E 


1 slot 


F 


2 slots 


G 


5 slots 


H 


10 slots 



At this point in the example, a total of 41 slots have been allocated and 159 slots 
remain free. 

• Now that all high level prioritization rules have been followed, the remainder 
of the slots will be allocated to honor as much of the remaining demand as 
possible. 

• Nodes B, C, F, G, and H still have "general demand" that has not been 
satisfied however Node H has already been allocated its maximum so it will 
get no more slots. 

• Doling out the remaining slots equally would yield 39.75 slots for each of the 
4 nodes. 

• Two of these four nodes don't need 39.75 slots; Node C needs 30 more and 
Node F needs 3 more. The demand for these two nodes is satisfied leaving 
126 slots left. 

• The two other nodes B and G may split up the remaining 126 slots 63 each but 
G only needs 55 more slots so its demand is satisfied leaving 71 slots 
remaining for Node B. 

The Final example allocation after satisfying as much remaining demand as possible: 



NODE 


Slots 


A 


1 slot 


B 


73 slots 


C 


40 slots 


D 


10 slots 


E 


1 slot 



33 



F 


5 slots 


G 


60 slots 


H 


10 slots 



A total of 200 slots have been allocated and 0 slots remain free. 

Example Allocation Summary: 

• Node B's demand was not completely satisfied because there were not enough 
slots available. 

• Node H's demand was not completely satisfied because it requested more than 
its maximum bandwidth. 

• Nodes A and E each received a slot even though they did not ask for one since 
their minimum rate constraint dictated that it must get at least one slot per 
frame. 

All the variables input to the DA decision component may be set up at system 
initialization (or when a network operator intervenes) with the exception of the real 
time demand information which is stored in a data structure and is constantly updated. 

During times of low network loading when the total munber of slots requested 
does not exceed the total number of slots available, the DA has no decision to make. 
The DA gives every node the number of slots it asked for, feathers them across the 
frame, and sends the BTP. The most significant benefit of this approach is realized 
when network loading is high and user nodes are requesting far more slots than the 
DA can give them. In fact, if the QoS service level queues are set deep enough at a 
given node, the node may request to use as many slots as possible. If the node is only 
able to transmit on one carrier at a time the node may request to use every slot in the 
frame of a given upstream. If the node has multiple transmitters, it may ask for every 



34 



slot on all upstream channels. When one or more nodes requests more than their 
share, DA enforces the rules. 

To enforce rate-shaping rules, the DA pre-computes the equivalent bit rate of 
one slot per frame. For example, if there are 10 bits per slot, 10 slots per frame, and 
10 frames per second, the bit rate of a single slot per frame is 100 bits per second. If a 
maximum rate rule of 200 bits per second is specified for a given node, DA ensures 
that no more than 2 slots are allocated to that node on any given frame. Similarly, if a 
minimum rate rule of 100 bits per second is specified for a given node, DA ensures 
that at least one slot per frame is allocated to that node on every frame. CIR rules are 
similar and may be considered variable minimum rate constraints. If a node has a 
CIR equivalent to 4 slots per frame, DA allocates at least 4 slots when the node 
demands 4 slots or more, however it need not allocate any slots if there is no demand. 

A new problem is introduced if a minimum rate rule of less than one slot per 
frame, for example 50 bits per second, is defined. This case requires that the DA 
process store state information about previous frames to ensure that the minimum rate 
rule is enforced. In this example, DA would assign one slot every other frame to 
achieve a 50 bit per second rate. Traditional TDMA systems use a concept of a 
superframe that spans N frames to potentially assign a slot every other frame to fiilfiU 
this requirement; superframe systems send a BTP once per superframe. However, DA 
advantageously establishes a relationship between slots and bits per second, and by 
saving state information about the frames that occurred in the past second, 
performance equivalent to that of superframe systems is realized. And, most 
importantly, since DA does not have to plan ahead N frames, it can change the BTP 
every frame which makes the system much more responsive to changing demand than 
a superframe based system. 



35 



Another benefit of storing state information from several previous frames is 
realized in cases where there is no choice, in a given frame, other than to make an 
unfair decision. Consider the case where there is only one upstream channel, 10 slots 
per frame, three user nodes A, B, and C, and no rate shaping rules set. Assume there 
is high demand at all three nodes so all of them are requesting 1 0 slots (the whole 
channel). A fair decision would be to allocate 3.333 slots per frame to each node. 
Since it is not possible to allocate 1/3 of a slot, DA allocates 4, 3, and 3 slots to nodes 
A, B, and C respectively. If demand remains high during the next frame, DA 
remembers that A got an unfair advantage during the last frame and it therefore it 
allocates 3, 4, 3 to nodes A, B, and C respectively. This rotation of the unfair 
advantage leads to faimess on average over time. 

An additional priority factor considered by the DA is the high level 
importance of different types of rules set by the network operator. These high level 
priorities are. Real Time data, minimum data rate, and CIR, in no particular order. 
The order of the high-level prioritizations chosen by an implementer of the DA 
process is not important for the algorithm to work, it is only required that some 
ordering be determined so that DA can "break ties" when necessary. 

An example of optimally feathering slots after all allocation decisions are 
made is illustrated in Figures 15-17. The feathering process begins by adding up the 
total number of slots available 1502 for a given frame. For example, if there are three 
upstream channels and 20 slots per frame, then the virtual frame 1502 is 60 slots long. 
Target slot assignments 1503 and slot assignments per node 1500 are calculated for 
every node which will equally space them throughout the virtual frame using a 
floating point number. For example, if 1 1 slots are to be allocated they would have to 
be allocated every 5.45 slots [0, 5.45. 10.9, 16.35, ....49.05, 54.5]. The numbers are 



36 



then converted to integers by taking the ceihng of each [0, 6, 1 1, 17, 22, 28, 33, 39, 44, 
50, 55]. Once a target allocation is chosen for all the nodes, they are allocated slots in 
the virtual frame 1504 one at a time starting with the node that will receive the most 
slots. The first slot in every target allocation is the 0th slot; see the example in Figure 
15 where the target allocation for Node 2 is [0, 8, 15, 23, 30, 38, 45, 53]. Since the 
0th slot is already occupied, and the interval between slots must be preserved for 
optimal feathering, every number is shifted up by one to produce [1,9, 16, 24, 31, 39, 
46, 54]. When these slots are entered into the virtual frame 1504, they all fit except 
for slot 39 which is already occupied. When this happens, the interval between slots 
must be violated and the feathering algorithm attempts to get as close as possible to 
the target by shifting one slot left, then one right, then two left until an open slot is 
found. 

Once all allocated slots have been placed into the virtual frame, it is separated 
into upstream channels with virtual slot 0 mapping to upstream 0, slot 0, virtual slot 1 
mapping to upstream 1, slot 0, and so on as illustrated. For example, as illustrated in 
Figure 16, the virtual frame is separated into three corresponding 20 slot frames, 1600, 
1602, and 1604, each corresponding to an upstream channel. 

It may be desirable to hop in frequency as infrequently as possible, in which 
case the slots may be reshuffled between upstream channels (along the frequency axis) 
to minimize hopping without changing the optimal feathering along the time axis. For 
example, as illustrated in Figures 16 and 17, the slots in 20 slot frames 1600, 1602, 
and 1604 are reshuffled to result in 20 slot intervals 1702, 1704, and 1706, 
respectively, which will result in reduced frequency hopping. 

An altemative embodiment includes an additional constraint in the DA and 
feathering algorithms to restrict slot assignments that would otherwise require a 

37 



frequency hop in consecutive slots. The additional constraint specifies a number of 
slots to be skipped when a frequency hop is scheduled. Further, these skipped slots 
are made available for allocation to other users so. they are not wasted because the 
constraint applies only to the node that is currently switching frequency. This 
altemative embodiment advantageously provides settling time to a transmitter's 
analog circuitry that must settle when the frequency is changed. 
Upstream Control Process 

Five upstream signal parameters are adjusted in real time for every user node 
in the network: Symbol Timing, Frequency, Power, Modulation, and FEC. Symbol 
timing guarantees that each node transmits in exactly the right symbol time within the 
TDMA frame. Frequency adjustments ensure that bursts stay centered in the 
allocated spectmm. Power, Modulation, and FEC adjustments are used to maintain an 
acceptable bit error rate even when the channel is experiencing a severe rain fade. 

At least two network features contribute to the responsiveness of the UC 
process. First, channel quality assessments are made every time a TDMA burst is 
received at the hub. The assessment consists of a Symbol Offset, Frequency Offset, 
and an SNR. These are attached to a trailer before the burst payload is forwarded to 
the central control server. By collecting information in real time, an accurate running 
average of all parameters for each node is maintained and used to make adjustments. 
A second feature is the synchronous BTP that notifies the demodulator on the hub 
receiver as to which user node generated the burst in each time slot. Since the hub 
receiver knows which modulation and FEC the nodes are using, it can reconfigure 
these settings in the demodulator on a slot-by-slot basis. 

Without synchronized BTP, switching modulation and FEC would still be 
possible, however, it would not be as efficient because latency would be added and. 



38 



potentially, data would be lost in the transfer. This latency would be introduced 
because the node would be notified of the change Ya second or more after the hub 
receiver would be notified. Therefore, the hub receiver would be trying to receive 
with the new modulation and / or FEC although bursts would still be in transit having 
been sent by the old modulation and / or FEC so they could not be correctly 
demodulated and decoded. Another problem that would occur without synchronous 
BTP is that modulation and FEC would have to either be statically mapped to specific 
slots in the firame or scheduled via messaging that would be subject to the latency 
problem described above. Such a static mapping would complicate any demand 
assignment algorithm since it would have an additional constraint that, for example, 
slot 0 on upstream 1 is using BPSK so that slot may only be assigned to nodes 
transmitting BPSK. No such restrictions exist when the synchronous BTP is 
employed. 

As mentioned above, the abihty to change modulation and FEC dynamically 
enables all available power to be used to increase throughput rather than wasting 
power to over come rain fades which happen infrequently. 

Figure 18 illustrates an exemplary embodiment of the Upstream Control 
Process. For each node, an appropriate modulation and FEC are chosen and 
adjustments to power, timing, and fi*equency offset are calculated and placed into a 
UCP message, by the Central Control Server, in step SI 800. UCP Messages are 
multiplexed into a downstream broadcast channel in the hub transmitter in step SI 802. 
The UCP Message is decoded by each user node, and new power, timing, and 
frequency offset parameters are immediately applied to a user node transmitter in step 
SI 804, while new modulation and FEC parameters are apphed on an upcoming fi-ame 
in SI 806. TDMA bursts are received in the hub receiver, where channel quality is 



39 



assessed and the assessment is appended to the burst before forwarding to the Central 
Control Server in step SI 808. In the Central Control Server, channel assessments are 
sorted by user node in step S1810, averaged with other bursts from the same user 
node, and stored in corresponding positions in an Assessment Data Structure 1850. A 
message is sent to the hub firmware every frame in step S 1812 to prepare for 
modulation and FEC changes. Synchronous BTP enables the hub receiver to 
accommodate changes in modulation or FEC on a slot by slot basis in step SI 814. 

Downstream Recovery Process 

SatelUte network system link budget constraints are often tighter on the 
downstream channel, thus, the downstream connection may be the first to drop out 
during a rain fade. Since there are a number of different types of RF equipment and a 
number of possible upstream / downstream data rate combinations, a universal 
network system should have the ability to adapt both directions to combat rain fade. 

An obstacle when considering an adaptive technique on the downstream 
channel is that the downstream channel is a point-to-multipoint connection as opposed 
to the multipoint-to-point connectivity in the upstream direction. Therefore, any 
special treatment of the signal to enable one node to receive it more clearly should not 
affect the ability of any of the other nodes to receive the same carrier. 

One way to solve this problem might be to enable the nodes to receive either a 
BPSK or QPSK signal. The hub could transmit a broadcast QPSK signal while clear 
sky conditions are present at all nodes and then it could switch to BPSK (and 
potentially a stronger FEC code) if any node reports a falling SNR. A problem with 
this solution is that by switching to BPSK, the data rate of the downstream channel is 
reduced; thus the whole network suffers because there is rain at a single user node site. 



40 



As the network size grows, the UkeUhood that at least one node is in a fade condition 
increases and the network will end up using the lower rate downstream a majority of 
the time. This negates any advantage of an adaptable downstream. 

Figures 19 (19A-19C) and 20 illustrate an exemplary embodiment of the 
Downstream Recovery process used to send "recovery" data to only the node which is 
suffering from a rain fade without changing the manner in which data is sent to the 
rest of the nodes in the network. In this example, all nodes in clear sky conditions 
will receive data along the primary path and all fading nodes will receive data that 
traverses the alternate path. Downstream data packets, intended for user nodes for 
which downstream recovery has been activated, are marked in the Central Control 
Server before forwarding, in step SI 900. In the hub transmitter step SI 950, marked 
packets are sent either to a primary or altemate modulation path by demux 1920 under 
control of selection step SI 922. Marked packets are sent to a separate layer 2 queue 
1902, passed through an altemate FEC encoder in step SI 908, and mapped for an 
altemate modulation in step S1910. Likewise, downstream data packets, intended for 
user nodes for which downstream recovery has not been activated, are not marked in 
the Central Control Server before forwarding, in step SI 900. In the hub transmitter, 
unmarked packets are sent to a separate layer 2 queue 1916, passed through a primary 
FEC encoder in step SI 904, and mapped for a primary modulation in step SI 906. 
These packets are then multiplexed in step SI 91 2, and modulated in step SI 91 4 at 
Layer 1 into the downstream broadcast channel, under the control of FEC block 
counter 1924 and altemate block enable step SI 91 8. A user node detects locally that 
its downstream signal is degraded or lost and automatically switches to the altemate 
FEC and modulation in step S2052. Each user node sends a remote status message 
containing the average downstream Signal to Noise Ratio (SNR) at a configurable 



41 



frame, in step S2054. Bursts received at the hub receiver are forwarded to the Central 
Control Server in step S2056. If reported downstream SNR is greater than a threshold 
value in step S2060, no action is taken. If SNR falls below the threshold in step 
S2060, the downstream recovery process is activated for that user node. In addition, 
if no bursts are received from a user node within a configurable time period, it is 
declared to be Out of Network and the downstream recovery process is activated for 
that user node in step S2058. 

Both the primary and alternate FEC codes map to the same number of PSK 
symbols. For example, the primary path may have 4096 bit FEC blocks which map to 
2048 QPSK symbols and the alternate path may have 2048 bit FEC blocks which map 
to 2048 BPSK symbols. This enables either block to be advantageously multiplexed 
interchangeably at Layer 1 just prior to the PSK modulator. An additional method 
employed is to orient the BPSK constellation on a 45 degree axis such that the signal 
energy is divided equally between the Inphase and Quadrature modulation phases. By 
doing this, the amplitude envelope of the signal does not change whether QPSK or 
BPSK is used so the signal "looks" the same to all gain stages in the system. This 
orientation also ensures that vahd symbol timing estimates may be made by the 
demodulators at each node regardless of the modulation because signal peaks and zero 
crossings will be seen on both phases. Further, since the BPSK constellation is a 
subset of the QPSK constellation, phase estimates (measurements of the degree of 
constellation rotation) are valid for either one. The only difference is that the sheer of 
the BPSK demodulator has a 180 degree decision boimdary instead of 90 degrees 
which results in a 3dB improvement in the ratio of Energy per bit transmitted to the 
average noise of the channel (Eb/No). To further improve the ability of nodes to 



42 



overcome a fade condition, the implementer selects a more powerful, lower rate FEC 
code for the alternate path than the code used in the primary path. 

An example of an embodiment of the claimed invention is illustrated in 
Figures 19B and 19C. In this example, by offsetting the BPSK constellation by 45 
degrees, the signal is distributed equally on the I and Q components and the points are 
common to the QPSK constellation. Therefore, the demodulators on user nodes in 
QPSK mode still make valid symbol timing, magnitude, and carrier phase 
measurements while a BPSK block is being received. 

Another advantage of the DR process is that the downstream packet- 
processing software need not be aware of which path (primary or altemate) the data is 
traversing. Each data packet is tagged with a Layer 2 MAC address so that each user 
node filters out packets intended for other nodes. The hub transmitter maintains a list 
of the MAC addresses of all nodes in DR mode and when a packet arrives for one of 
those nodes, it is routed to the altemate path. Nodes are added and deleted from this 
list by the central control server as they enter and exit fade conditions, a feature which 
is enabled because the size of the FEC block payload (a Layer 1 property) and the 
data packet size (a property of Layer 2 and higher) are decoupled. This is difficult or 
impossible in systems without this decoupling, like the DVB satellite broadcast 
standard, because variable sized data packets are segmented (and later reassembled) 
into segments that fit into the FEC payload. In such systems, DR would be much 
more cumbersome because the higher layers would have to account for which nodes 
were in DR mode and which were not. 

In addition, although DR has been discussed so far only in terms of a primary 
and altemate modulation and encoding, respectively, the present invention also 
applies to altemative embodiments having multiple levels of altemative modulation 



43 



and encoding. For example, the robustness of the Unk may be adjusted by using 
combinations of modulation and coding schemes, where the robustness of a link 
generally increases with a greater amount of energy-per-bit as compared to system 
noise, and more redundancy in the code employed, as discussed in Sklar, B, "Digital 
Communications Fundamentals and Applications," Prentice Hall, 1988, the entire 
contents of which being incorporated herein by reference. 

Figure 21 illustrates a computer system 2101 upon which an embodiment of a 
user node, or hub, of present invention may be implemented. The computer system 
2101 includes a bus 2102 or other communication mechanism for communicating 
information, and a processor 2103 coupled with the bus 2102 for processing the 
information. The computer system 2101 also includes a main memory 2104, such as 
a random access memory (RAM) or other dynamic storage device (e.g., dynamic 
RAM (DRAM), static RAM (SRAM), and synchronous DRAM (SDRAM)), coupled 
to the bus 2102 for storing information and instructions to be executed by processor 
2103. In addition, the main memory 2104 may be used for storing temporary 
variables or other intermediate information during the execution of instructions by the 
processor 2103. The computer system 2101 further includes a read only memory 
(ROM) 2105 or other static storage device (e.g., programmable ROM (PROM), 
erasable PROM (EPROM), and electrically erasable PROM (EEPROM)) coupled to 
the bus 2102 for storing static information and instructions for the processor 2103. 

The computer system 2101 also includes a disk controller 2106 coupled to the 
bus 2102 to control one or more storage devices for storing information and 
instructions, such as a magnetic hard disk 2107, and a removable media drive 2108 
(e.g., floppy disk drive, read-only compact disc drive, read/write compact disc drive, 
compact disc jukebox, tape drive, and removable magneto-optical drive). The storage 



44 



devices may be added to the computer system 2101 using an appropriate device 
interface (e.g., small computer system interface (SCSI), integrated device electronics 
(DDE), enhanced-IDE (E-IDE), direct memory access (DMA), or ultra-DMA). 

Figure 22 illustrates an example of an embodiment of a user node 304 
according to the present invention. In this example, the user node 304 includes a 
computer 2200 connected to a plurality of network interfaces 2202. The computer 
2200 is a general purpose microprocessor based device including volatile and non- 
volatile storage and standard interfaces. The network interfaces 2202 provide access 
to 10/100 Ethernet, RS-232, V.35, analog telephony, optical networks, gigabit 
Ethernet, USB, IEEE 1394, wireless LAN, and other networks. The computer 2200 is 
also connected to optional additional processing peripherals 2204 to provide 
additional processing resources such as encryption/decryption engines, data or voice 
compression devices, content addressable memory devices, video codecs and other 
functions that may be implemented in programmable logic or custom logic devices 
configured for custom processing purposes. In addition, the computer 2200 is 
connected through a bus 2212 to a plurality of RF transmitters 2208, RF receivers 
2210, and expansion peripheral connections 2206. The RF transmitters and receivers 
provide a communication link to the satellite 300, and from there to the hub 302 and 
to other user nodes 304. The expansion peripheral connections 2206 provide optional 
connectivity to other devices compatible with the bus 2212 such as storage devices, 
wireless LAN devices, sound or video cards, lottery systems, point of sale devices, 
public kiosks, information signs, public internet terminals, bus protocol converters, 
and other peripherals. Altematively, the network interfaces may connect directly to 
the bus 2212 instead of connecting to the computer 2200. 

Numerous modifications and variations of the present invention are possible in 



45 



light of the above teachings. It is therefore to be understood that within the scope of 
the appended claims, the invention may be practiced otherwise than as specifically 
described herein. 



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